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<DIV><FONT face=Arial size=2>I'm running Asterisk SVN-trunk-r51353... for some
reason even if i set nat=yes in the sip.conf for a device when i do a show sip
peers it shows N for nat. Is this a bug or am i doing somthing wrong
here. I'm basically having a problem right now where i can call in/out of
asterisk and talk fine using the phone but if i call another SIP phone
registered to the same asterisk server it rings but when i pickup both ends i
cant hear or say anything through them. Not sure if this is related to the
NAT issue... i think it may be?</FONT></DIV>
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<DIV><FONT face=Arial size=2>my setup is...</FONT></DIV>
<DIV><FONT face=Arial size=2>Asterisk is on a public ip</FONT></DIV>
<DIV><FONT face=Arial size=2>2 polycom601 phones on a private
network</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>here's a sip debug...</FONT></DIV>
<DIV><FONT face=Arial size=2><A
href="http://channels.debian.net/paste/5164">http://channels.debian.net/paste/5164</A></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>~Shaun</FONT></DIV>
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